In recent years, due to widespread use of the Internet and IP phones, a method for performing facsimile communication using an IP network as a communication pathway is being established. A call control protocol, such as Session Initiation Protocol (SIP), for generating, changing, and disconnecting a session for real-time communication is being standardized. Also, audio communication on a VoIP network that performs communication by assuming a modem signal tone as a voice with the use of a VoIP (Voice over Internet Protocol) has started to spread. Also, T.38 protocol for converting a G3 facsimile (ITU-T recommendation T.30) signal into an IP message on an IP network and transmitting the IP message in real time is recommended, and an internet facsimile apparatus that conforms to ITU-T recommendation T.38 is also on the market. Facsimile communication using this T.38 protocol has a bandwidth that is able to be smaller than that of the audio communication on VoIP network, which needs a voice band, and does not need to modulate data into a modem signal, resulting in an advantage of allowing high-speed communication.
Also, an IP-PBX (Internet Protocol Private Branch eXchange) is spreading as an apparatus for performing circuit switching of IP phone terminals in a private IP network. This IP-PBX is an apparatus for realizing an extension telephone network using IP phones on in-House LAN, and uses the call control protocol such as SIP and H.323 to perform call control of the IP phones and the like. A real-time internet facsimile apparatus uses the SIP to obtain an IP address of a communication partner via this IP-PBX, and performs P2P communication with the communication partner. Generally, in the case of performing T.38 communication, a user designates the media type of SIP as “application” or “image” so as to perform communication, and designates the media type of SIP as “audio” so as to perform telephone talk.
IP-PBXs that can perform T.38 communication include an apparatus that needs to start operating with the media type designated as “audio” at the time of establishment of a session using SIP, even in the case of performing T.38 communication, and then performs facsimile communication by the media type being switched to “image”.
Also, a Subscriber Line Interface Circuit (SLIC) module is provided as a technology for establishing an IP phone. This module performs A/D conversion processing and telephone line emulation using an audio codec such as G.711. With the SLIC, it is possible to connect an analog telephone to an IP network without the use of a telephone line, thus providing a function that emulates a telephone line, such as generation of a call signal and various types of tone signals of a telephone line (telephone exchange).
FIG. 3 is a diagram illustrating a conventional incoming call sequence.
In the case where an IP phone using a SLIC receives a session establishment request (INVITE signal) for establishing a session of the “audio” media type from a sender side, the IP phone generates a call signal for an analog telephone connected to the SLIC to make a sound, and the generated call signal causes the ringer of the analog telephone to make a sound. Then, when an operator responds to the analog telephone, the SLIC detects that the analog telephone is in an off-hook state, stops the call signal, and transmits a success response (200 OK) to the sender side, and the sender returns ACK. Accordingly, an audio session is established, and the operator is placed in a talk state.
Japanese Patent Laid-Open No. 2011-114672 gives a conventional example for controlling incoming calls of an IP phone. According to this document, in a telephone network, an incoming call is first received by an interactive voice response, and a ringer tone is not rung immediately after the reception of the incoming call. In the interactive voice response, for each connection, a sender is notified of an authentication tone obtained by superimposing random numbers subjected to random number processing each time and background tones one on top of the other, as an authentication tone needed for the connection, and an incoming call is rung only for an incoming call in which a correct response value is received, and otherwise, no incoming call is rung.
In the following case, it is assumed that not only when telephone talk is performed but also when T.38 communication is performed, an IP-PBX that needs to start operating with a media type designated as “audio” in the case of a session establishment request (INVITE signal) using SIP is used. The communication terminal on a receiver side cannot determine, based on the session establishment request (INVITE signal), whether the incoming call is for telephone talk or T.38 communication. Therefore, a failure may occur such that an incoming call for calling an operator is made even in the case of an incoming call for T.38 communication, and thus a no-ring status for incoming calls becomes impossible in a facsimile apparatus.
Also, there is a real-time internet facsimile apparatus that supports, in addition to the T.38 communication function, an IP telephone function, and a function for performing audio communication on a VoIP network. Such a facsimile apparatus needs an incoming call control processing for selecting the most appropriate communication method from the received session establishment request, in order to support, also on the IP network, functions of a conventional G3 facsimile apparatus including a telephone function, which has, in addition to the telephone function, a plurality of receiving modes.